This research is directed toward improved signal processing schemes to aid people with sensorineural impairments. The two classes of schemes to be studied are multichannel amplitude compression (for listeners with reduced dynamic range) and pitch-invariant frequency lowering (for listeners with negligible hearing at high frequencies). The proposed work on amplitude compression will focus on the evaluation of a 16-channel computer-controlled system. Evaluations will be conducted when the system is adjusted to restore the entire set of normal equal-loudness contours for tones, and when these adjustments are modified to take account of the loudness relations among sounds more speechlike than tones, the degraded intensity resolution caused by compression, the spread of masking that occurs at high levels, and the speech confusions that are observed. For comparative purposes, we will also evaluate a linear amplification system that simulates the best previous compression system. The proposed work on frequency lowering differs from prior (generally unsuccessful) efforts in the choice of lowering schemes and in the attention paid to training. Very roughly, the lowered speech signals will be similar to those that would be produced by a talker with normal vocal cords but a giant vocal tract, and have spectograms that are altered by a continuous monotonic transformation of the frequency axis. Initial work includes a study that employs real-time, pitch-invariant, uniform lowering and focuses on the development of training procedures; a study that employs off-line processing to achieve pitch-invariant nonuniform lowering and focuses on the effects of different amounts and types of lowering; and the development of a real-time system for the study of pitch-invariant, nonuniformly lowered speech. Later work will evaluate pitch-invariant, nonuniformly lowered speech with real-time processing and extensive training. In all cases, we will attempt to understand the limitations of the signal-processing schemes and model the perception of the transformed speech. BIBLIOGRAPHIC REFERENCES: Frazier, R.H., Samsam, S., Braida, L.D., and Oppenheim, A.V., "Enhancement of Speech for Adaptive Filtering," Proc. 1976 IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, Philadelphia, May 1976.